Guys, you all disappoint me. The point of higher sampling rates is capturing at best quality possible, regardless if the quality improvement to 16/44.1 is only theoretical. The point is: you can always dowsample later. Really, all your shiny, biting and ironic arguments would have made sense ten years ago but in present times itīs completely superfluous: nowadays we have plenty of HDD space, more than capable hardware, fast processing (even 32/384.000 can be edited and processed very fast if I may be allowed to indulge in that exaggeration fantasy) and extremely fast internet connections. It just doesnīt make sense to use 16/44.1 or lossy compression nowadays and even less sense to promote it as the only format worthwile.

Did it occur to you that for example the Grammy foundation and the Deutsche Grammophon are doing backups of their precious analogue tapes with 24/192 before the tapes degrade and are lost? They are doing this because for example the IASA recommends it. They also donīt care if the quality is better or not, they just want to capture everything. They apparently do it according to the motto 'Better safe than sorry'. Are you calling them stupid or deluded? They obviously know the times they are in and behave accordingly, they exploit the technical possibilities they have and donīt stand still. Which is what has been missing here since a few years now. You guys are still doing comparisons of lossy formats at 96 kBit/s! Why? The world wonīt need them for much longer. Furthermore, you make fun of people who donīt share your opinion. Granted, some of them deserve it. But for a few years now everyone who promotes something like higher samplerates, 24 bit, hell, anything else that deviates from established norms is treated like a little dumb child. You have become close minded people, living in their own secluded world that is becoming smaller and smaller every day.

This forum clearly has lost its edge, I really feel that the world has turned - but without you.

I will no longer participate here (youīll applaud it, Iīm sure) and I know that sentences will be put out of context. If there would be a delete-button for my profile Iīd use it.

The audio world disappoints me, solving non-problems by adopting the increases in sample rate and bit depth that Moor's Law brings almost for free, while ignoring the possibilities of decent surround sound, the high cost and/or poor performance of transducers, and the lousy sound in most consumers' homes.

DG should use more than 192kHz - people have already made great use of the bias signal recovered from tapes to stabilise wow and flutter. Better make sure they're capturing it faithfully to allow this processing in the future.

Of course, if there is ever any objective evidence that 192kHz brings audible benefits for home listening, HA would be extremely interested.

Cheers,David.

P.S. I've found hobbyists who insist on transferring vinyl at 192kHz while using a $200 turntable. I wonder where they got the idea that high sampling rates were so important? Not from HA.

Guys, you all disappoint me. The point of higher sampling rates is capturing at best quality possible, regardless if the quality improvement to 16/44.1 is only theoretical. The point is: you can always dowsample later.

I love the steps forward high resolution movies come to me now on Blu-Ray and HDTV. I did some step ups on my TV hardware over the time and always enjoyed it getting better and better. I love to watch HD-TV, especially sport events, it simply makes fun.At night me old fart puts on his 3D glasses and slaughters enemies in 3D on a Playstation 3. Looking good on a recent 47" LED TV only wasting 60Watts of power.

But i still canīt hear, as end user the stated benefits of Hires music against its same version properly downsampled. Believe me my audio system is not to shabby and allows native Hires playback.I donīt question to archive the raw capture of a recording somewhere in the best possible, recent technology. I am just fed up that on some forums it is a given all these Hires formats sound always better.If someoene wants to tell me i do not because i missed the world turning i have to wonder...

People record with 96 or 192 kHz simply because they can and because they believe it sounds better.

Try a few VST compressors at 44100 Hz and then on a project using 192000 Hz and listen to the difference in attack and release. this does not mean that the percieved quality of the recorded audio is better but some plugins sound a lot better at higher sample rates. Regards.

I've been part of this past year fixing most of the plugins of my program* so that they don't perform differently at different sample rates. There's no plausible reason why a compressor would act that much different at different sampling rates. The slope should increase accordingly. (Of course, by the nature of a compressor, a higher samplerate allows to control better the envelope, but we're talking about a couple of samples better, not something that could clearly be heard).

It just doesnīt make sense to use 16/44.1 or lossy compression nowadays and even less sense to promote it as the only format worthwile.... You guys are still doing comparisons of lossy formats at 96 kBit/s! Why? The world wonīt need them for much longer. Furthermore, you make fun of people who donīt share your opinion.

I completely agree with you on the fun-making part. Such people don't contribute anything but rather deteriorate a forum's reputation. But I strongly disagree with you on the rest.

Digital archiving of historical recordings is a completely separate issue, where data rate doesn't matter. So yes, if the analog tape creates distortion up to 48 kHz, digitize at 96 kHz, why not? But why at 24 or 32 bits? You won't capture any more information than with 16 (or 14) bits per sample. And, luckily, at HA many know a thing or two about information theory which most of the world's population doesn't.

Sorry to disappoint you, but already nowadays you hear more compressed than lossless audio in your daily life. Digital TV, radio, phone, basically every animation or stream on the Internet use it. Give me an application other than physical media (CD, Blu-Ray, etc.) and private FLAC/ALAC collections where lossless audio is being distributed/used by more than a tiny minority. That's why we test things like 96 kbps: to see how low you can go with lossy compression and still achieve excellent (but of course not transparent) quality, and to make other people aware of our findings. It's a hard truth, but I think we are reaching a point where lossless audio is becoming unimportant. The industry is e.g. pushing digital radio to channel bit-rates far below 100 kbps stereo! Which is ridiculous if you ask me, but that's how it is. So there are much more important things to worry about than whether to use 48 kHz or higher sampling rates.

Edit: Thought about it some more... Of course nowadays we would also want to archive e.g. some contemporary amateur cell-phone videos with "historical value". Isn't it more important to get the crappy cell-phone video and sound right in the next phone generation than to be able to preserve it in 24/96?

Guys, you all disappoint me. The point of higher sampling rates is capturing at best quality possible, regardless if the quality improvement to 16/44.1 is only theoretical. The point is: you can always dowsample later. Really, all your shiny, biting and ironic arguments would have made sense ten years ago but in present times itīs completely superfluous: nowadays we have plenty of HDD space, more than capable hardware, fast processing (even 32/384.000 can be edited and processed very fast if I may be allowed to indulge in that exaggeration fantasy) and extremely fast internet connections. It just doesnīt make sense to use 16/44.1 or lossy compression nowadays and even less sense to promote it as the only format worthwile.

Did it occur to you that for example the Grammy foundation and the Deutsche Grammophon are doing backups of their precious analogue tapes with 24/192 before the tapes degrade and are lost? They are doing this because for example the IASA recommends it. They also donīt care if the quality is better or not, they just want to capture everything. They apparently do it according to the motto 'Better safe than sorry'. Are you calling them stupid or deluded? They obviously know the times they are in and behave accordingly, they exploit the technical possibilities they have and donīt stand still. Which is what has been missing here since a few years now. You guys are still doing comparisons of lossy formats at 96 kBit/s! Why? The world wonīt need them for much longer. Furthermore, you make fun of people who donīt share your opinion. Granted, some of them deserve it. But for a few years now everyone who promotes something like higher samplerates, 24 bit, hell, anything else that deviates from established norms is treated like a little dumb child. You have become close minded people, living in their own secluded world that is becoming smaller and smaller every day.

This forum clearly has lost its edge, I really feel that the world has turned - but without you.

I will no longer participate here (youīll applaud it, Iīm sure) and I know that sentences will be put out of context. If there would be a delete-button for my profile Iīd use it.

Jeez, such dramatics!

High SR and high bit depth have their place at the recording and production end -- the utility of 24 or 32bit processing is a given here, and 2bdecided even noted one of the most compelling (and little-cited) reasons to use super-high SR for archiving master tapes: it allows later use of software that eliminates flutter (Plangent processing).

However, they're largely bling at the delivery format end....i.e. the media that the consumer buys for the mere purpose of *listening*.

Anyway, back to reality: if you are making high quality recordings, and some of the people purchasing your high quality recordings want to pay you $10 extra for a 192kHz version, why on earth wouldn't you make one available? As long as it doesn't make the quality worse, and doesn't cost you more than the financial return, it's really not a problem if people want to pay more for no tangible benefit.

I think it should be quite clear to anyone here that it's of no audible benefit what-so-ever, but it may create an excuse (that the accountants will accept) to create better (re-)masters. Which will then be used for the 44.1kHz version that can now be bought for a bargain price. Everyone's a winner.

Not necessaraly so....

A general rule of measurements is that accuracy and measurement time are related. Low measuring times means low accuracy and high accuracy means long measuring times.So there comes a time that while you think you increase the amount of information (increase sampling rates), you are actualy decreacing the amount of information. Again information theory explains this all.

There are two ways to realise this is misleading...

1) It's the RMS noise that typically increases as sampling frequency increases. The dB/Hz noise doesn't. So the noise level within the audio band remains roughly constant as sample rate is increased. When we didn't have enough bits to out-do the real world, the in-band noise fell as the sample rate increased.

2) No ADCs or DACs run at 44.1kHz natively. They run at a higher rate internally. The 96kHz output is no less accurate than the 48kHz output - both are usually derived from the same higher rate version internally. The 96kHz version can't be worse than the 48kHz version.

1) It's the RMS noise that typically increases as sampling frequency increases. The dB/Hz noise doesn't. So the noise level within the audio band remains roughly constant as sample rate is increased. When we didn't have enough bits to out-do the real world, the in-band noise fell as the sample rate increased.

Is there any particular point you'd like to make? Or do you agree with 2Bdecided's (essentially correct) response.

Sure, namely a part of the 4th paragraph in the article:

QUOTE

There is also a tradeoff between speed and accuracy. Conversion at 100MHz yield around 8 bits, conversion at 1MHz may yield near 16 bits and as we approach 50-60Hz we get near 24 bits. Speed related inaccuracies are due to real circuit considerations, such as charging capacitors, amplifier settling and more. Slowing down improves accuracy.

And in general itīs the shorter time you take to measure, the higher the tolerances.Iaw: You canīt have the cake and eat it.

And in general itīs the shorter time you take to measure, the higher the tolerances.

Don't forget the fact that at a higher sampling rate you have a higher bandwidth the noise can spread into. You may get more noise but how much of it will go into the band-of-interest? The interesting question is: How does the power spectral density of the measurement noise change in the band-of-interest with a different sampling rate?

Edit: Oh, I just saw that 2B basically touched the same question. Power spectral density is usually measured in dB/Hz if I recall correctly.

The downside is that we don't get proper surround. Though some people are still quietly working on that too.

I'm intrigued by this part...how does industry support for silly sample rates keep us from getting proper surround?

There's only so much effort, marketing, messaging etc - and the parts of those aimed at "better sound" are mostly going the wrong way IMO. Resources, and audiophile's attention, are not infinite.

Or to put it another way, they've got to sell something. They've chosen to sell imagined improvements, rather than real ones, a) because it's easier, and b) because plenty of people who should know better haven't called "foul".

As others have pointed out, that's contradictory - it makes the point that virtually all sampling now uses hundreds of kHz or MHz...

QUOTE

For example, most front ends of modern AD (the modulator section) work at rates between 64 and 512 faster than a basic 44.1 or 48KHz system. This is 16 to 128 times faster than 192KHz. Such speedy operation yields only a few bits. Following such high speed low bits intermediary outcome is a process called decimation, slowing down the speed for more bits. There is a tradeoff between speed and accuracy. The localized converter circuit (few bits at MHz speeds) is followed by a decimation circuit, yielding the required bits at the final sample rate.

...but then claims...

QUOTE

Sampling audio signals at 192KHz is about 3 times faster than the optimal rate. It compromises the accuracy which ends up as audio distortions.

So there's a point in the ADC where the input is a many kHz few bits signal. This can be converted to 48kHz, 96kHz, or 192kHz. In each case, the quality in the audio band is the same. Downsampling to 192kHz rather than 48kHz cannot introduce more audio distortions.

There is no technical point to using 192kHz - I agree with the paper in this respect - but this specific point "It compromises the accuracy which ends up as audio distortions" - or the idea that there's more noise in the audio band when sampling at 192kHz vs 48kHz, is wrong.

Cheers,David.

P.S.

QUOTE

There is also a tradeoff between speed and accuracy. Conversion at 100MHz yield around 8 bits, conversion at 1MHz may yield near 16 bits and as we approach 50-60Hz we get near 24 bits. Speed related inaccuracies are due to real circuit considerations, such as charging capacitors, amplifier settling and more. Slowing down improves accuracy.

This is true, buta) it doesn't tell you anything about the noise in the audio band, andb) in the context of 48kHz vs 192kHz, we're talking about the exact same analogue electronics (capacitors etc) - it's only the digital (mathemataical!) downsampling that's adjusted.

As others have pointed out, that's contradictory - it makes the point that virtually all sampling now uses hundreds of kHz or MHz...

QUOTE

For example, most front ends of modern AD (the modulator section) work at rates between 64 and 512 faster than a basic 44.1 or 48KHz system. This is 16 to 128 times faster than 192KHz. Such speedy operation yields only a few bits. Following such high speed low bits intermediary outcome is a process called decimation, slowing down the speed for more bits. There is a tradeoff between speed and accuracy. The localized converter circuit (few bits at MHz speeds) is followed by a decimation circuit, yielding the required bits at the final sample rate.

...but then claims...

QUOTE

Sampling audio signals at 192KHz is about 3 times faster than the optimal rate. It compromises the accuracy which ends up as audio distortions.

So there's a point in the ADC where the input is a many kHz few bits signal. This can be converted to 48kHz, 96kHz, or 192kHz. In each case, the quality in the audio band is the same. Downsampling to 192kHz rather than 48kHz cannot introduce more audio distortions.

There is no technical point to using 192kHz - I agree with the paper in this respect - but this specific point "It compromises the accuracy which ends up as audio distortions" - or the idea that there's more noise in the audio band when sampling at 192kHz vs 48kHz, is wrong.

Cheers,David.

P.S.

QUOTE

There is also a tradeoff between speed and accuracy. Conversion at 100MHz yield around 8 bits, conversion at 1MHz may yield near 16 bits and as we approach 50-60Hz we get near 24 bits. Speed related inaccuracies are due to real circuit considerations, such as charging capacitors, amplifier settling and more. Slowing down improves accuracy.

This is true, buta) it doesn't tell you anything about the noise in the audio band, andb) in the context of 48kHz vs 192kHz, we're talking about the exact same analogue electronics (capacitors etc) - it's only the digital (mathemataical!) downsampling that's adjusted.

What matters is how fast the actual information is gathered, at 192kHz its faster than at 44.1kHz. Therefore the tolerances at 192kHz will be higher than at 44.1kHz.

So?

What do you think is preferable?(a) use a sampler/quantizer at 44.1 kHz which adds Gaussian white noise with a variance of v(b) use a sample/quantizer at 192 kHz with adds Gaussian white noise with a variance of 3v (three times as high)?

I'll tell you: (b) is preferable because the noise's power spectral density is 1.6 dB/Hz lower compared to (a). It means: less noise in the audible band.

I was under the impression that 24 bits offers headroom (for lack of a better word) for destructive editing operations?

All of what I say here exclusively focuses on recording. Of course for editing/processing, it makes sense to work with 24 or more bits, just like you would work with a 24- or 32-bit image before creating an 8-bit GIF or PNG of it. Also note in response to punkrockdude that (in my wishful thinking) I assumed that nowadays, plug-in developers know what they're doing when it comes to audible aliasing and other distortions. From what people tell me here, it seems that this is not always the case...

It has little to do with editing. People record at higher bit depths so that they don't have to worry about running out of headroom during the recording process from unexpected transients. You can easily leave 20dB of headroom without sacrificing SNR, at least from a mathematical perspective.

IOW, you don't need to be as careful about your levels with 24-bit as you do recording in 16-bit. If the levels were optimized at 16-bit upon recording vinyl then recording at 24-bit would be completely unnecessary, since 16-bit is more than adequate in presenting material sourced from vinyl.

It has little to do with editing. People record at higher bit depths so that they don't have to worry about running out of headroom during the recording process from unexpected transients. You can easily leave 20dB of headroom without sacrificing SNR, at least from a mathematical perspective.

If you goal is 20 dB headroom, 16 bits can almost always provide it.

A really quiet room with musicans in it rarely provides a maximum signal that is more than 66 dB above the acoustic noise floor, and you can observe this given reasonbly quiet mics. That's about 30 dB below the 93-96 dB of nominal dynamic range that is available with 16 bits. If you noise-shape the quantization, then there is even more dynamic range than 96 dB with just 16 bits.

I've been part of this past year fixing most of the plugins of my program* so that they don't perform differently at different sample rates. There's no plausible reason why a compressor would act that much different at different sampling rates. The slope should increase accordingly. (Of course, by the nature of a compressor, a higher samplerate allows to control better the envelope, but we're talking about a couple of samples better, not something that could clearly be heard).